The reason to use sample rates for PCM audio over 96kHz is so that you can have a much larger file size, that takes up more cpu to process, while making it sound worse. The reason people go for doing this is because of the obviously understandable belief that "bigger is better" - something that counter intuitively is simply not necessarily true with PCM audio. Unfortunately some manufacturers attempting to push their hardware add to the disinformation around these things.

Looking further at the feature set of this app - DDP creation, batch processing, DSD recording and playback, bundled set of plugins - it looks like a really good deal for those in the need of a mastering app.

while I'm absolute a BIG fan of 192kHZ when capturing mastering from tape , I see reasons for DSD and higer sample rates. specially when your final master/mix format is not in the digital domain, it's nice to have some options in capturing/distributing it in different digital formats. If it makes money, it makes money !
and some people just use music to listen to their stereo's and for me there is nothing wrong with a listeners point of view, which sometimes get's ridiculed by "pro's"

Yes.... I can't share them at this time...I'll ask for permission when the project is done.

I think this often depends also on the converters. The Meitner ADC8 MK IV was great at DSD with 128 FS. But the PCM output of the same unit at 24/96 wasn´t at the same level.

Now IMHO there are some PCM converters which are better at 24/192 than the Meitner ADC at DSD 128FS.

Do we need more than 192k?
I think the step from 48k to 96k is obvoiusly an improvment. The difference from 96k to 192k is smaller but with great releases on adequate systems I hear an advantage to 96k.
More than 192k? Probably no benefit.

Is it because of higher rate DSD "needing" higher rate dxd, double 384 ? Just asking.

As I understand it, DSD records the 1-bit delta-sigma signal directly rather than decimating that data down to a 96kHz (or other) PCM sample rate. However, the 1-bit sample rate is up around 2MHz - so I don't know what we actually have here (maybe I need to research DSD more!).

It appears from the Sound It! website that 768kHz is the actual PCM sample rate it supports.

Not all VST plugins support even 192kHz - so the plugins that come with this software must either be modified versions or they are actually downsampling at the plugins (defeating the purpose).

while I'm absolute a BIG fan of 192kHZ when capturing mastering from tape , I see reasons for DSD and higer sample rates. specially when your final master/mix format is not in the digital domain, it's nice to have some options in capturing/distributing it in different digital formats. If it makes money, it makes money !
and some people just use music to listen to their stereo's and for me there is nothing wrong with a listeners point of view, which sometimes get's ridiculed by "pro's"

anyway )) lot's of great plugs for the $

The capture for me at 11.2, sounds different compared to the pcm capture.. I am using the merging hapi converter....When you utilize different A/Ds the outcome of the resulting file can sound different.I'm sure everyone has their favorite pitch and capture A/D and D/A. And over time,and with some testing, you may find something else that may tickle your fancy...

Most people won't need a sample rate as high as 768 kHz, but it is not totally useless. Higher sample rates simplify the task of accurately reconstructing both the phase and magnitude of very high frequencies. Sometimes multiple samples can be combined in such a way that there is an improvement in the SNR. It is also possible that one might acquire audio that happens to have very high sample rate. For example, say you use a specialized machine to digitize a fragile 140 year old recording. It is also conceivable that one might need the extra precision for some scientific reason.

I think the real reason, though, is that the sample rate needs to be high enough for 99.9% of all potential customers. Also, adding support for 768 kHz is often a trivial task for the software developers.

Did you read the paper of Wescott? There is no academic conflict with what Shannon says.

The first sentence in this paper states what everybody knows.

"The assertion made by the Nyquist-Shannon sampling theorem is simple: if you have a signal that is perfectly band limited to a bandwidth of f0 then you can collect all the information there is in that signal by sampling it at discrete times, as long as your sample rate is greater than 2f0."

The next statement often is neglected.

"The difficulty with the Nyquist-Shannon sampling theorem is that it is based on the notion that the signal to be sampled must be perfectly band limited. This property of the theorem is unfortunate because no real world signal is truly and perfectly band limited. In fact, if a signal were to be perfectly band limited—if it were to have absolutely no energy outside of some finite frequency band—then it must extend infinitely in time."

Another 2 statements in this paper are:

"Nyquist didn’t say that if you are sampling at rate N that you could use an anti-aliasing filter with a cutoff at frequency f = N/2."

"If you have the freedom to set the sampling rate, this filtering effect means that you’ll need to make a trade-off between a low sampling rate and the complexity of your anti-alias filtering. It is not uncommon for systems to have sampling rates that are seemingly quite high, just to simplify the task of designing the anti-alias filters."

I cannot see an academic contradiction in this paper to that what Nyquist and Shannon says.

Did you read the paper of Wescott? There is no academic conflict with what Shannon says.

The first sentence in this paper states what everybody knows.

"The assertion made by the Nyquist-Shannon sampling theorem is simple: if you have a signal that is perfectly band limited to a bandwidth of f0 then you can collect all the information there is in that signal by sampling it at discrete times, as long as your sample rate is greater than 2f0."

The next statement often is neglected.

"The difficulty with the Nyquist-Shannon sampling theorem is that it is based on the notion that the signal to be sampled must be perfectly band limited. This property of the theorem is unfortunate because no real world signal is truly and perfectly band limited. In fact, if a signal were to be perfectly band limited—if it were to have absolutely no energy outside of some finite frequency band—then it must extend infinitely in time."

Another 2 statements in this paper are:

"Nyquist didn’t say that if you are sampling at rate N that you could use an anti-aliasing filter with a cutoff at frequency f = N/2."

"If you have the freedom to set the sampling rate, this filtering effect means that you’ll need to make a trade-off between a low sampling rate and the complexity of your anti-alias filtering. It is not uncommon for systems to have sampling rates that are seemingly quite high, just to simplify the task of designing the anti-alias filters."

I cannot see an academic contradiction in this paper to that what Nyquist and Shannon says.

Hello,

I know the problem of the numeric is the passage of the mathematic object (they are perfect) to the electronic reality.

Many argue 44.1KHz is enough with proper oversampling on both ends. Doesn't quite match what I've heard, with my ears agreeing with the above statement that there's the most benefit at 96K with a slight improvement at 192. But there are so many factors involved, quality of the DAC, ADC, clocking, DSP, oversampling, mastering, etc. But the math says one can get accurate reconstruction of a 20KHz bandwidth limited signal at 44.1KHz. But whether frequencies above 20K add to the audio experience or not is often argued.

This is really all about what the filters, which vary in construction from both device to device and sample rate to sample rate, introduce below 20kHz. Adding to the excitement is that different people are more and less sensitive to the artifacts. This subject makes for endless arguments whenever it is oversimplified. I think I've literally heard every sample rate beat every other sample rate when using some specific gear. In the end, I need to use whatever sounds best using my own gear.